This invention relates to a digital-analog converter, and more particularly, to a digital-analog converter suitable for use in converting a digital audio signal into an analog audio signal.
In compact disc players (CD players) or digital tape recording/playback devices (DAT devices), it is required that musical signals expressed in digital form be converted into analog signals prior to output.
As shown in FIG. 39, a commonly employed digital-analog converter (hereinafter referred to as a "DA converter") for playing back music includes a digital current converter 1 for converting digital data DT, which is inputted at a certain sampling period, into a direct current I.sub.o, a current-voltage converter 2 for converting the current I.sub.o into a voltage S.sub.D (see FIG. 40), and for holding the voltage, each time a sampling pulse P.sub.s is generated, and a low-pass filter 3 for forming the output voltage S.sub.D into a continuous, smooth analog signal S.sub.A, which is the output of the filter 3. The current-voltage converter 2 includes a switch SW having a movable contact changed over by the sampling pulse P.sub.s. When the movable contact is switched to a contact a, as shown in FIG. 39, an integrator is formed to generate the voltage S.sub.D, which conforms to the current I.sub.o. When the movable contact is switched to a contact b, a holding circuit is formed to hold the voltage S.sub.D.
The foremost problems encountered in the DA converter for music playback are the precision with which the digital data is converted into a current value, the speed at which the conversion is made and phase distortion caused by the low-pass filter.
The problems of conversion precision and conversion speed have largely been solved by higher speed LSI's and advances in trimming techniques. Though phase distortion ascribable to the low-pass filter can be mitigated by employing a digital filter, phase distortion cannot be eliminated completely so long as the filter is an integral part of the structure.
FIG. 41 is useful in describing phase distortion. FIG. 41(a) illustrates an original audio signal waveform 5a, a 1 KHz component waveform 5b, and an 8 KHz component waveform 5c. FIG. 41(b) illustrates an audio signal waveform 6a outputted by the low-pass filter 3 (FIG. 39), a 1 KHz component waveform 6b, and an 8 KHz component waveform 6c. It will be understood from these waveforms that, due to the delay in the phase of the 8 KHz component, the output audio signal 6a is different from the original audio signal 5a, and that this phase distortion becomes particularly pronounced at high frequencies. Thus, the presence of the low-pass filter results in a major deterioration in sound quality.
As shown in FIG. 42, the low-pass filter output when a pulsed signal is applied to the filter is sluggish at a leading edge 7a and oscillates at an envelope portion 7b and trailing edge 7c. Consequently, when a musical signal exhibiting a large impulse variation is applied to the low-pass filter, sound quality changes greatly and there are times when even the rhythmical sense of the musical signal differs.